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00:18.44 | pancho_jay | Hello all! |
00:18.51 | pancho_jay | I have problems with asterisk11 and odbc |
00:18.58 | pancho_jay | I am trying to use realtime configuration for sip peers |
00:19.06 | pancho_jay | but asterisk shows me a lot of messages like these: |
00:19.12 | pancho_jay | [Mar 5 20:06:31] WARNING[19260]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: ERROR: column "regseconds" is of type bigint but expression is of type character at character 149; |
00:19.21 | pancho_jay | I don't know why Asterisk is trying to save regseconds field as string (instead of integer) |
00:19.36 | pancho_jay | I am using PostgreSQL 9.5 and Ubuntu 14.04 64bits |
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02:45.46 | jdub_35 | hello... is anyone there? |
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03:18.32 | jdub_35 | anyone there? |
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04:16.12 | kunwon1 | hi, using asterisk 13.5.0 - with res_hep and res_hep_rtcp - and using ngrep to monitor outgoing traffic. If I unload res_hep_rtcp, there is zero traffic to my sip capture server. if i leave it loaded, i get rtcp HEP packets but nothing else (no registration, call info, etc) - any tips on how to fix? |
05:24.00 | drmessano | What is the power supply voltage on a Polycom phone? Like a 330/331 |
05:24.04 | drmessano | 24v? |
05:26.40 | kunwon1 | drmessano: yes |
05:28.49 | drmessano | Thanks.. guess I need to work out the current draw now |
05:30.40 | kunwon1 | drmessano: the factory power supply supplies 0.5A |
05:30.51 | drmessano | Awesome.. thank you! |
05:31.39 | drmessano | I wonder if you can run 24v to them PoE |
05:31.50 | drmessano | Im guessing not |
05:32.06 | kunwon1 | they run with any modern PoE switch |
05:32.54 | drmessano | I know that much |
05:33.13 | drmessano | Looks like they dont work with 24v on the ethernet cable |
05:33.23 | drmessano | SO I will need a splitter |
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08:33.06 | XATRIX | Hi, why when i change my SIP port 5060->6060 my SIP client stops to make\receive calls ? |
08:33.29 | XATRIX | I can register on PBX but, i can 't make calls. debug info says : NAT type: Symmetric |
08:33.51 | XATRIX | But if i switch back to 5050 ,everything goes ok |
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09:03.04 | drmessano | why are you changing it? |
09:03.06 | drmessano | nm |
09:30.33 | snadge | 5060 is just too mainstream ;) |
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09:44.04 | drmessano | Yeah |
09:44.16 | drmessano | Gotta keep out the evil hackers |
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09:55.23 | hypknight|Away | is now away - Reason : Auto-Away (Away from Keyboard for 60 minutes) |
10:10.54 | wdoekes | hypknight|Away: you're wasting a lot of peoples time by getting attention for that auto-away message in multiple channels |
10:11.02 | wdoekes | don't do that |
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14:39.18 | pancho_jay | Hello all! |
14:39.55 | pancho_jay | I am trying to setup Asterisk with realtime support for SIP peers, but it isn't working |
14:40.10 | pancho_jay | Asterisk shows me a lot of warning messages |
14:44.31 | pancho_jay | WARNING[25800]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: ERROR: column "regseconds" is of type bigint but expression is of type character at character 143; |
14:45.09 | pancho_jay | I don't know why Asterisk is trying to save regseconds field as a string. Regseconds should be an integer |
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15:59.00 | voip | thank you so much |
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18:14.24 | kline | is csipsimple on android known to not work with asterisk, or is the debian package for asterisk just junk? |
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18:14.57 | kline | im following the tutorial on the wiki with the debian packages, and i can make sip extensions and dial in, but all my calls are silent |
18:18.35 | kline | I'm pretty sure the calling itself is working - if I change the Wait() parameter, I see matching behaviour in the call |
18:20.13 | WIMPy | Any NAT involved? |
18:20.17 | WIMPy | ~sipnat |
18:20.17 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
18:22.13 | kline | shouldnt be, we're on the same lan. asterisk is running on a vm but the interfaces are bridged rather than natted |
18:22.35 | kline | I'll have a read in any case |
18:22.44 | WIMPy | Maybe the client is by default configured for NAT support. |
18:23.06 | kline | the AO link btw is 404 |
18:23.20 | WIMPy | Thw what? |
18:23.37 | kline | http://www.aocomputing.net/?p=3 |
18:23.55 | kline | from the ~sipnat snippet |
18:24.15 | WIMPy | Oh, still :-( |
18:36.42 | kline | ok, so im still not 100% convinced theres a nat involved, but csipsimple says that the local nat type is "port restricted" |
18:37.15 | WIMPy | Well, it thinks there is. |
18:37.53 | WIMPy | That's enough to cause trouble. Turn it off or configure Asterisk as if the client was behind NAT. |
18:37.59 | kline | yeah |
18:41.55 | kunwon1 | anyone successfully using res_hep? I have enabled that and res_hep_rtcp and only the latter is sending UDP packets to my capture server, there is nothing being generated by res_hep |
18:42.28 | kunwon1 | i updated to 13.7.2 and the problem persists |
18:44.46 | kline | WIMPy, ok, thanks, that sorted it at least for hello-world |
18:54.21 | kunwon1 | looks like res_hep only works with pjsip, not chan_sip |
18:57.08 | WIMPy | Sounds likely. |
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20:30.25 | julius | hey |
20:32.54 | julius | by chance i got my hands on something called a "easybox", when creating a SIP account in asterisk is there any way to test that without having to attach the box to the phone line? |
20:33.44 | julius | i mean like a with a softphone, call the box and have the call redirected to asterisk |
20:33.55 | julius | does that make sense at all? |
20:34.14 | WIMPy | Not sure. can you even configure that thing? |
20:35.05 | WIMPy | And what kind of Box is it? |
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